Conferencing is a mature billion dollar industry with a user base that continues to grow. The business benefits of remote meetings are well proven and despite poor user experiences, conferencing is seen as a necessity in most corporations. Despite over 20 years of conference calls, users still have to endure lengthy and complicated conference set-ups and audio calls that are often poor quality.
I have over 10 years experience in the conferencing domain and believe that WebRTC has the potential to transform how conferencing service providers (CSPs) deliver and monetise their services. Here’s why:-
The vast majority of remote meetings are still accessed through PSTN
Web-conferencing promised to make virtual meetings simpler, smoother and more engaging but it never really delivered. End users need decent audio quality and a simple user experience but current Web conferencing solutions just don’t provide that. Indeed, web to audio integration delivers poor audio quality and still requires browser plug-ins (Flash player), which are a pain to manage within corporate IT environments. In most cases, Flash player is used as an “audio patch” to HTML’s lack of audio capabilities; Flash simply wasn’t designed to carry real-time voice. Whilst there are success stories, web conferencing is still largely limited to document sharing with around 90% of audio conferences still being led through PSTN access. Adobe Connect today has a technical edge thanks to its ability to customize the flash plug-in to the Connect features but audio quality (not to mention cost) remains an issue for mass adoption.
WebRTC will allow Web-conferencing to really take-off
WebRTC delivers real-time native audio over the web using HTML5 with no plug-in download required. Early comparisons with Flash reveal that WebRTC can deliver faster connection times, lower audio latency, and better audio quality than existing PSTN and Web conferencing solutions.
WebRTC, not only makes conferencing smoother, but it opens up a wide range of opportunities to create new and innovative applications which enhance the meeting experience.
Like any new and potentially disruptive technology, WebRTC makes existing capabilities simpler to deliver and more economical. Here is a quick snapshot of the areas where I think it is relevant for conferencing services:
Improved user experience and data integration capabilities
Traditional conferencing requires complex DTMF line reconciliation or “hard to charge for” PSTN dial-out features. Audio identification on Web apps is straightforward with WebRTC connections, and by leveraging user identification, audio recordings can become real assets. For example, metadata can be gathered to analyse the activity and behaviour of participants during calls and CSPs can use that data to increase the business value of their services.
HTML5 doesn’t just enable WebRTC. It offers additional and improved bi-directional data communication technologies between the browser and the server. Improved data and real-time communication integration is possible; a specific example being more efficient active speaker notifications, which can greatly improve the user experience.
Typical business audio meetings have a fairly small number of participants making a good case for audio mixing on the client side rather than a remote server. This delivers a lower latency and lower infrastructure costs whilst keeping bandwidth requirements acceptable. For meetings with a greater number of participants, there is a good case for a hybrid client/server mixer based upon the conference size.
Improved QoS and enhanced user experience – Opus and 3D spatial conferencing
Opus is a mandatory to implement (MTI) codec for WebRTC, delivering up to full-band stereo – a huge improvement compared to G.711 which delivers narrow-band quality on PSTN.
More importantly, Opus support enables the transmission of 3D spatial audio which creates the impression that voices are coming from unique points in space, to bring a real life audio experience to the conference call. Symonics contributed to the Opus standard and clearly see an opportunity with 3D spatial conferencing with WebRTC. And the recent announcement that BT Conferencing is deploying 3D spatial audio technologies in their network confirms the market demand for this type of solution.
Whilst a 3D audio mixer is required to deliver spatial conferencing with WebRTC, it is a great example of server-side WebRTC services which deliver a differentiated user experience.
Faster innovation and large developer community
Higher service availability and flexibility to address specific needs
WebRTC does not impose a signalling protocol between clients and servers, and sessions can be implemented to include resiliency features. PSTN and SIP did not allow much flexibility because they enforced their own view of service availability. Now it is up to the application developer to define the resiliency mechanisms that address the conferencing service providers’ specific needs.
How can service providers monetise WebRTC conferencing?
Service providers should not see WebRTC as a threat to existing revenues; it could actually enable higher margins on existing services as well as potentially creating new revenue streams.
The current PSTN access brings additional costs with local PTT interconnects which could be reduced through the use of WebRTC. Adoption will also mean lower infrastructure costs given the fact that most CSPs today have IP-enabled infrastructures accessed through expensive TDM gateways. This results in comparative WebRTC port costs being lower than the PSTN equivalent, enabling CSPs to strengthen profit margins or create new competitive cost models.
WebRTC also offers the opportunity for CSPs to monetize their core mixing infrastructure by exposing it to innovative Web developers who will provide WebRTC access to end users. This would allow addressing high margin market verticals not being looked at by CSPs today.
Conference Service providers need to act now!
WebRTC can fix Web-conferencing. Up for grabs are better QoS, improved user experience, lower costs and opportunities for differentiation in a crowded market.
There is a window of opportunity for CSPs to add TelCo value to WebRTC-based applications. For this to happen, CSPs need to embrace the dynamic and innovative web developer community and provide access to their core conferencing infrastructure. This will place web developers in a position to provide custom conferencing solutions for market verticals such as education, government and healthcare.
For CSPs to succeed and monetise WebRTC, they need to be the core enabler of new, user centric applications and services. If they don’t embrace WebRTC now they risk seeing the emergence of new players leveraging the technology to provide better collaboration tools at a lower cost. Ultimately users will have the final say, but if WebRTC delivers better QoS, as well as a simpler and richer service at the same price or lower, the choice should be easy.
What do you think? Will traditional conferencing Service Providers embrace WebRTC? What should they be doing now to improve conferencing services on the web?
Romain Testard is a senior conferencing product manager at NetDev with over 10 years experience in developing and managing conferencing and collaboration services for Tier 1 service providers. Romain has broad experience in the Voice and Web collaboration space coupled with a deep technical understanding of SIP and WebRTC technologies. Romain holds an MSc in Telecoms Signal Treatment as well as an MBA from “Université du Sud” in France.